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Full text of "Implementation of Amplitude Modulation on Software Defined Radio"

Implementation of Amplitude Modulation on 
Software Defined Radio 

K.Satyanarayana 1 , Madhekar Suneel 2 , Ravi Ranjan 2 

Indian Institute of Technology-Madras, Chennai-600036, eel0b014@ee.iitm.ac.in 

2 PGAD, Defence Research and Development Organisation, Hyderabad-500058, cybersuneel@gmail.com 

and raviranjan222@gmail.com 



Abstract — An amplitude modulation transceiver design is 
presented, using a software defined radio platform. Model 
based design is used to implement the various transmitter 
and receiver blocks. The wireless communication of audio 
signals is demonstrated. 

Index Terms — Software radio, wireless communication, am- 
plitude modulation, demodulation. 



I. Introduction 

This papers presents a design of an amplitude modu- 
lation (AM) transceiver. The modulation scheme used is 
double sideband modulation. The demodulation is carried 
out using an envelope detector. The entire implementation 
is carried out on a software defined radio (SDR) platform. 
The SDR platform used is the Lyrtech Small Form Factor 
(SFF) platform. Model based design is used as the de- 
sign method. This method relies on Matlab®, Simulink®, 
Texas Instruments® Code Composer Studio® and Xilinx® 
blocks. 

II. Amplitude Modulation 

Amplitude modulation is defined as a process in which 
the amplitude of the carrier wave A c cos(27rf c t) is varied 
about a mean value, linearly with the baseband message 
signal m(t), where, A c is the carrier amplitude, f c is the 
frequency of the carrier signal and t is time. The amplitude 
modulated signal is given by 



s(t) =Ac[l + k a m (£)] cos (2nf c t) , 



(1) 



where, k a is the amplitude sensitivity of the modulation 
scheme. In frequency domain, this can be expressed as 



S(f) = -y[*(/-/c) + *(/ + /c)] + 
^[M(/-/ c ) + M(/ + / c )]. 



(2) 



III. Software Defined Radio 

SDR is a wireless technology created to improve interop- 
erability between different wireless networks, field radios, 
and devices. With this technology, we can create multi- 
mode, multi-band, and multi-functional wireless devices 
and network equipment that can be dynamically recon- 
figured, enhanced, and upgraded through software updates 

® All trademarks are property of their respective owners 



and hardware reconfiguration. An SDR consists of three 
main modules Fig 2, namely, data conversion module, 
digital signal processing module and RF section module 
[2]. A picture of the SDR platform used to carry out this 
work is shown in Fig 1. 



B 


— ~ 


' m 


» 




i 


1 




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• 
• 


' . B 


,,,_ ^ 




Hv.ii 


I 


- 


L 


i I 


iV "" 


i 





Fig. 1 : SDR platform with mobile phone as audio source 



A. Digital signal processing module 

The digital signal processing (DSP) module contains a 
six core TMS320DM6446DSP processor, and a Xilinx® 
Virtex-4® FPGA. The processor is clocked at 594 MHz. 
The processor and the FPGA form the main signal process- 
ing devices of the radio platform. The DSP module also 
contains an audio codec. 

B. Data conversion module 

This module contains two analog input channels with 
14 bit, 125 MSPS analog to digital converters (ADC). 
The ADC used is a Texas Instruments® ADS5500 [4]. 
This module also contains two analog outputs driven by a 
dual channel 16-bit, 500-MSPS digital to analog converter 
(DAC). The DAC used is a Texas Instruments® DAC5687 
[5]. Besides, there are two external clock inputs and on- 
board clock synthesizer. 

C. RF module 

This module has a radio frequency transceiver with 
frequency range 1.6 GHz to 2.2 GHz. The intermediate 
frequency bandwidth is selectable between 5 MHz and 



20 MHz. The maximum gain provided is 22 dB. The RF certain other platform- specific configuration blocks. The 
section sensitivity is -110 dBm at a signal to noise ratio of FPGA model receives the samples from the DSP, the FPGA 
10 dB and a signal bandwidth of 1 kHz. model is shown in Fig 4. 





\7 




^Z 




RF module 






RFin 


RF out 









31 



Data conversion module 



ADC 



FPGA 



DAC 



31 



Digital signal processing module 



DM6446 DSP 



FPGA 




90 degree phaser 



ADACMaster III 



Fig. 4: FPGA model showing AM modulation 

The received signal is first upsampled to 125 MSPS, the 
upsampling to 125 MSPS in Fig 4 is shown in figure 5. 



Fig. 2: Schematic of the modules 



IV. Description of AM transmitter 

The description of the AM transmitter is in two parts, 
namely, the DSP model and the FPGA model. The DSP 
model is implemented in the DSP, and acts as an interface 
between the audio signal source, and the FPGA. With 
the on-board audio codec as the signal source, a DSP 
model is shown in Fig 3. The audio codec is configured to 



Da ta type conv ersion 



^H 



Audio code I/O 



VPSSBus 




El 



Tx spectrum 



CPU Percentage 



RF Module Control 



Fig. 3: DSP transmitter model 

generate output samples at 24.414 kHz. A Video Processing 
Subsystem (VPSS) bus is used to communicate the received 
samples to the FPGA. The VPSS bus provides 32-bit 
interface between DSP to FPGA and FPGA to DSP. The 
VPSS bus is provided with the Video Processing Front 
End (VPFE) directs the signal to DSP and the Video 
Processing Back End (VPBE) directs the signal to the 
FPGA. The VPFE is clocked at 75 MHz and the VPBE 
is clocked at 37.5 MHz. The DSP model also contains 
blocks to configure the RF section, the audio codec and 
the ADACMaster-III data conversion module, apart from 



( 1 ) — ► In1 Out1 



x10 Interpolator X 8 Interpolator x 8 Interpolator! x8 Interpolator 



Fig. 5: Upsampling to 125 MSPS 

The upsampling by 8 block (x8 interpolator) is imple- 
mented as shown in Fig 6. The other upsampling blocks 
in Fig 5 are implemented in a similar manner. 



*► cast — ►lb] ^t 




Convert Slicel Reinterpret 



Fig. 6: Upsampling by a factor 8 

The upsampled signal is amplitude modulated using a 
1 MHz carrier waveform. The carrier is generated using a 
direct digital synthesis (DDS) block as shown in Fig 7. 

The modulated signal is given to the DAC, for upcon- 
version to the Intermediate Frequency (IF). In the DAC, 
single side band modulation mode is enabled. This IF 
signal is then given to the RF module for upconversion 
and transmission. The transmission frequency used in the 
present work is 1700 MHz. 



reinterpret 



mod index 



:=1 -DK7 1 



reinterpret! 



IMhzDDS 



cmult 



nilt l». 



mult 



Add 



Out 




cast — Wa:b] ■ 



Convert Slicel Reinterpret 
Terminatorl 



Down Sample2 



-H-3 



Fig. 7: Amplitude modulation 



Fig. 10: Downsampling by a factor 8 



V. Description of AM receiver 

The transmitted signal propagates over the air, and is 
received by the receive antenna of the RF module. The 
RF module down-converts it to the IF of 30 MHz. This IF 
signal is fed to one of the ADC inputs of the ADACMaster- 
III data conversion module. The digitized signal is provided 
as input to the signal processing module. In the signal 
processing module, the description is again in two parts, 
namely the DSP model and the FPGA model. FPGA model 
is shown in figure 8. 



configuratiorri 

Downconversion 



& D ownsamplin c 



% 



In1 



sgn 




Threshold 



Slice 




Constant6 



dO 



In1 Out1 



Out1 



env FIR2 



Mux 




Fig. 11: Envelope Detector 



VPSS bus 



Fig. 8: FPGA model showing AM demodulation 

In the FPGA model the signal is first downconverted by 
30 MHz , as shown in Fig 9 and downsampled by factor 
8 as shown in Fig 10. This down-sampled signal is fed 



\fti 



ffWflff 



Downconversion_Filter 



30 Mhz DDS 




— J z" Jnterp ► b 



. Delay for Retiming Reinterpretl 



Fig. 9: Downconversion 

to the input of an envelope detector block, shown in Fig 
11. The envelope detector demodulates the AM signal. The 
demodulated signal is further down-sampled to 24.414 kHz 
by downsampling blocks, as shown in Fig 12. 

The down-sampling blocks in Fig 12 are implemented 
in a similar manner as in Fig 10. 



This signal is communicated from the FPGA, 
another VPSS bus to the DSP . 



using 



In the DSP model this signal is fed to the DC offset 
block. This block adjusts the DC of the signal. This 
adjusted signal is fed to the audio codec with samples at 
24.414 kHz, as shown in Fig 13. 





In1 


Out1 




In1 


Out1 




In1 


Out1 


In1 


w 


w 



Decimatorl by 8 



Decimator2 by £ 



Decimator3 by 10 



Fig. 12: Downsampling to 24.414 kHz 



*!J!J|.' CMD File Generator Set bitstream Lyrlech priority manager Audio codec C 




Data Type Conversion 



*^ 



Audio Codec I/O 



*H 



14000 




















12000 




















10000 
3000 


















(vW 


<D 

~ 6000 

§_4000 
^ 2000 
















"V 


\l 























■2000 




















4000 




















-6000 

























0.2 

:8567 


0.4 


0.6 


0.8 1 1.2 

Time{mins) 


1.4 


1.6 


1.8 2 



Fig. 13: DSP receiver model 



Fig. 16: Received signal in time domain 



VI. Results 

The results are taken from Simulink® so there is a 
time delay between the transmitted baseband signal and the 
received signal. There is a DC component in the received 
spectrum as an effect of envelope detection. 

Fig 14 and Fig 15 are baseband signals (audio) from the 
Tx wave and Tx spectrum in the Fig 3 




Fig. 14: Signal from audio codec in time domain 




CD ; 




50 100 150 200 250 300 350 400 450 500 

le 8567 Frequency(mHz) 



Fig. 17: Received signal in frequency domain 



References 

[1] Rodger H. Hosking. Software Defined Radio Handbook, Pentek Inc, 

New Jersey, 2011. 
[2] Jeffrey H. Reed. Software Radio: A Modern Approach to Radio 

Engineering, Pearson Education, New Jersey, 2002. 
[3] Simon Haykin. Communication Systems, Wiley, New York, 3rd ed, 

1994. 
[4] Texas Instruments® ADS5500 datasheet available at http://www.ti. 

com/lit/ds/symlink/ads5500.pdf 
[5] Texas Instruments® DAC5687 datasheet available at http://www.ti. 

com/lit/ds/symlink/dac5687 .pdf 



200 250 300 350 400 

Frequency(mHz) 



Fig. 15: Signal from audio codec in frequency domain 



Fig 16 and Fig 17 are baseband signals (audio) from 
the Rx wave and Rx spectrum in the Fig 13. The received 
signal was played back on a speaker and reproduced well.